global_s Posted September 14, 2012 Report Share Posted September 14, 2012 Hello, I have setup 2 domain in my Snom blue Originally I setup an inbound trunk to go directly to a conference room and it worked. Then I setup the same inbound trunk to the second domain and suddenly, it stopped working for both. So I removed the second trunk, made the first global and check the result. Calls are received by the pbx, I can hear the welcome message to the conference, but DTFM are not received, so I can't enter a room. If I call from an extension everything works. I disable the only trunk from being global with no success. Not even bringing it back to the original configuration solves the issue. I recreated the trunk with no luck Please help Here's my log from a test call. I ended it because the system couldn't understand my digits. [5] 2012/09/15 00:03:18: SIP Rx udp:81.201.83.45:5060: INVITE sip:8877@sips14.s.tw SIP/2.0 Call-ID: IUP7KEWC4BDSHIFRHS65MA27HI@81.201.83.45 CSeq: 102 INVITE From: "39333333333" <sip:39333333333@voxbone.com>;tag=19899 To: <sip:8877@sips14.s.tw> Via: SIP/2.0/UDP 81.201.83.45:5060;branch=z9hG4bKce77d15175f9ec73a2dcd59c38df0a0b Max-Forwards: 69 Content-Type: application/sdp Contact: <sip:39333333333@81.201.83.45:5060;transport=udp> User-Agent: Vox Callcontrol Content-Length: 240 v=0 o=root 20488 20488 IN IP4 81.201.83.53 s=session c=IN IP4 81.201.83.53 t=0 0 m=audio 11516 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv [8] 2012/09/15 00:03:18: Allocating for call port 11, SIP call id IUP7KEWC4BDSHIFRHS65MA27HI@81.201.83.45 [9] 2012/09/15 00:03:18: UDP(IPv4): Opening socket on 0.0.0.0:10084 [9] 2012/09/15 00:03:18: UDP(IPv4): Opening socket on 0.0.0.0:10085 [9] 2012/09/15 00:03:18: UDP(IPv6): Opening socket on [::]:10084 [9] 2012/09/15 00:03:18: UDP(IPv6): Opening socket on [::]:10085 [5] 2012/09/15 00:03:18: Identify trunk (IP address match) 12 [9] 2012/09/15 00:03:18: Resolve 256: aaaa udp 81.201.83.45 5060 [9] 2012/09/15 00:03:18: Resolve 256: a udp 81.201.83.45 5060 [9] 2012/09/15 00:03:18: Resolve 256: udp 81.201.83.45 5060 [5] 2012/09/15 00:03:18: SIP Tx udp:81.201.83.45:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 81.201.83.45:5060;branch=z9hG4bKce77d15175f9ec73a2dcd59c38df0a0b From: "39333333333" <sip:39333333333@voxbone.com>;tag=19899 To: <sip:8877@sips14.s.tw>;tag=8828815886 Call-ID: IUP7KEWC4BDSHIFRHS65MA27HI@81.201.83.45 CSeq: 102 INVITE Content-Length: 0 [7] 2012/09/15 00:03:18: Set packet length to 20 [6] 2012/09/15 00:03:18: Call-leg 11: Sending RTP for IUP7KEWC4BDSHIFRHS65MA27HI@81.201.83.45 to 81.201.83.53:11516, codec not set yet [8] 2012/09/15 00:03:18: Incoming call: Request URI sip:8877@sips14.s.tw, To is <sip:8877@sips14.s.tw> [8] 2012/09/15 00:03:18: Call from a trunk 12 [8] 2012/09/15 00:03:18: Trunk Voxbone@sipmsk.of.tw has country code not set, area code not set [9] 2012/09/15 00:03:18: Incoming: formatted From is = "39333333333" <sip:39333333333@voxbone.com;user=phone> [9] 2012/09/15 00:03:18: Incoming: formatted To is = <sip:8877@sips14.s.tw;user=phone> [9] 2012/09/15 00:03:18: Incoming: formatted URI is = sip:8877@sipmsk.of.tw;user=phone [8] 2012/09/15 00:03:18: To is <sip:8877@sips14.s.tw;user=phone>, user 31, domain 1 [8] 2012/09/15 00:03:18: From number 39333333333 is configured as cell phone for extension 43402@sips14.s.tw, display name Francesco Silvestri [8] 2012/09/15 00:03:18: To user 8877 [8] 2012/09/15 00:03:18: Send call to extension is not set. Route the call based on global number 8877 [5] 2012/09/15 00:03:18: Global trunk Voxbone@sipmsk.of.tw sends call to 8877 in domain sips14.s.tw [8] 2012/09/15 00:03:18: From number 39333333333 is configured as cell phone for extension 43402@sips14.s.tw, display name Francesco Silvestri [8] 2012/09/15 00:03:18: Set the To domain based on To user 8877@sips14.s.tw [8] 2012/09/15 00:03:18: Call state for call object 12: idle [7] 2012/09/15 00:03:18: Call port 11: set_codecs for IUP7KEWC4BDSHIFRHS65MA27HI@81.201.83.45 codecs "", codec_preference count 7 [8] 2012/09/15 00:03:18: Call state for call object 12: connected [8] 2012/09/15 00:03:18: Play space10, caching false [8] 2012/09/15 00:03:18: call port 11: state code from 0 to 200 [7] 2012/09/15 00:03:18: Set packet length to 20 [9] 2012/09/15 00:03:18: Call port 11: update_codecs for IUP7KEWC4BDSHIFRHS65MA27HI@81.201.83.45: adding codec pcmu/8000 to available list [9] 2012/09/15 00:03:18: Call port 11: update_codecs for IUP7KEWC4BDSHIFRHS65MA27HI@81.201.83.45: Other side does not support codec pcma/8000 [9] 2012/09/15 00:03:18: Call port 11: update_codecs for IUP7KEWC4BDSHIFRHS65MA27HI@81.201.83.45: Other side does not support codec g722/8000 [9] 2012/09/15 00:03:18: Call port 11: update_codecs for IUP7KEWC4BDSHIFRHS65MA27HI@81.201.83.45: Other side does not support codec g729/8000 [9] 2012/09/15 00:03:18: Call port 11: update_codecs for IUP7KEWC4BDSHIFRHS65MA27HI@81.201.83.45: Other side does not support codec g726-32/8000 [9] 2012/09/15 00:03:18: Call port 11: update_codecs for IUP7KEWC4BDSHIFRHS65MA27HI@81.201.83.45: Other side does not support codec gsm/8000 [9] 2012/09/15 00:03:18: Call port 11: update_codecs for IUP7KEWC4BDSHIFRHS65MA27HI@81.201.83.45: codec_preference size 7, available codecs size 2 [6] 2012/09/15 00:03:18: Call-leg 11: Codec pcmu/8000 is chosen for call id IUP7KEWC4BDSHIFRHS65MA27HI@81.201.83.45 [5] 2012/09/15 00:03:18: set codec: codec pcmu/8000 is set to call-leg 11 [9] 2012/09/15 00:03:18: Resolve 257: aaaa udp 81.201.83.45 5060 [9] 2012/09/15 00:03:18: Resolve 257: a udp 81.201.83.45 5060 [9] 2012/09/15 00:03:18: Resolve 257: udp 81.201.83.45 5060 [5] 2012/09/15 00:03:18: SIP Tx udp:81.201.83.45:5060: SIP/2.0 200 Ok Via: SIP/2.0/UDP 81.201.83.45:5060;branch=z9hG4bKce77d15175f9ec73a2dcd59c38df0a0b From: "39333333333" <sip:39333333333@voxbone.com>;tag=19899 To: <sip:8877@sips14.s.tw>;tag=8828815886 Call-ID: IUP7KEWC4BDSHIFRHS65MA27HI@81.201.83.45 CSeq: 102 INVITE Contact: <sip:8877@174.16.25.5:5060;transport=udp> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: snomONE/4.5.0.1075 Delta Aurigids Content-Type: application/sdp Content-Length: 249 v=0 o=- 512677875 512677875 IN IP4 174.16.25.5 s=- c=IN IP4 174.16.25.5 t=0 0 m=audio 10084 RTP/AVP 0 101 a=rtpmap:0 pcmu/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=rtcp-xr:rcvr-rtt=all voip-metrics a=sendrecv [6] 2012/09/15 00:03:18: Call-leg 11: Codec pcmu/8000 is chosen for call id IUP7KEWC4BDSHIFRHS65MA27HI@81.201.83.45 [5] 2012/09/15 00:03:18: set codec: codec pcmu/8000 is set to call-leg 11 [5] 2012/09/15 00:03:18: SIP Rx udp:81.201.83.45:5060: ACK sip:8877@174.16.25.5:5060;transport=udp SIP/2.0 Call-ID: IUP7KEWC4BDSHIFRHS65MA27HI@81.201.83.45 CSeq: 102 ACK From: "39333333333" <sip:39333333333@voxbone.com>;tag=19899 To: <sip:8877@sips14.s.tw>;tag=8828815886 Via: SIP/2.0/UDP 81.201.83.45:5060;branch=z9hG4bK81ebc6e5fe28edb493f1bf6c44b4a660 Max-Forwards: 69 Contact: <sip:39333333333@81.201.83.45:5060;transport=udp> User-Agent: Vox Callcontrol Content-Length: 0 [5] 2012/09/15 00:03:20: SIP Rx udp:188.40.65.170:27010: REGISTER sip:174.16.25.5 SIP/2.0 Via: SIP/2.0/UDP 188.40.65.170:27010;branch=z9hG4bK0bf989e7;rport From: <sip:924385@174.16.25.5>;tag=as5ba10c39 To: <sip:924385@174.16.25.5> Call-ID: 5d05e6ae2909f0c12505dc292323cada@188.40.65.170 CSeq: 209 REGISTER User-Agent: PBX Max-Forwards: 70 Expires: 3600 Contact: <sip:Snomone@188.40.65.170:27010> Event: registration Content-Length: 0 [9] 2012/09/15 00:03:20: Resolve 258: aaaa udp 188.40.65.170 27010 [9] 2012/09/15 00:03:20: Resolve 258: a udp 188.40.65.170 27010 [9] 2012/09/15 00:03:20: Resolve 258: udp 188.40.65.170 27010 [5] 2012/09/15 00:03:20: SIP Tx udp:188.40.65.170:27010: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 188.40.65.170:27010;branch=z9hG4bK0bf989e7;rport=27010 From: <sip:924385@174.16.25.5>;tag=as5ba10c39 To: <sip:924385@174.16.25.5>;tag=8e52c4dad1 Call-ID: 5d05e6ae2909f0c12505dc292323cada@188.40.65.170 CSeq: 209 REGISTER Content-Length: 0 [5] 2012/09/15 00:03:20: SIP Rx udp:81.26.212.150:6060: OPTIONS sip:2.228.101.190:24942 SIP/2.0 Via: SIP/2.0/UDP 81.26.212.150:6060;branch=0 From: sip:pinger@81.26.212.150;tag=7f105f66 To: sip:2.228.101.190:24942 Call-ID: 4f7b936-3b5e9ec8-4d604e4@81.26.212.150 CSeq: 1 OPTIONS Content-Length: 0 [9] 2012/09/15 00:03:20: Resolve 259: aaaa udp 81.26.212.150 6060 [9] 2012/09/15 00:03:20: Resolve 259: a udp 81.26.212.150 6060 [9] 2012/09/15 00:03:20: Resolve 259: udp 81.26.212.150 6060 [5] 2012/09/15 00:03:20: SIP Tx udp:81.26.212.150:6060: SIP/2.0 200 Ok Via: SIP/2.0/UDP 81.26.212.150:6060;branch=0 From: <sip:pinger@81.26.212.150>;tag=7f105f66 To: <sip:2.228.101.190:24942>;tag=e5b876add8 Call-ID: 4f7b936-3b5e9ec8-4d604e4@81.26.212.150 CSeq: 1 OPTIONS Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Content-Length: 0 [5] 2012/09/15 00:03:20: SIP Rx udp:81.26.212.150:6060: OPTIONS sip:2.228.101.190:24942 SIP/2.0 Via: SIP/2.0/UDP 81.26.212.150:6060;branch=0 From: sip:pinger@81.26.212.150;tag=8f105f66 To: sip:2.228.101.190:24942 Call-ID: 4f7b936-4b5e9ec8-4d604e4@81.26.212.150 CSeq: 1 OPTIONS Content-Length: 0 [9] 2012/09/15 00:03:20: Resolve 260: aaaa udp 81.26.212.150 6060 [9] 2012/09/15 00:03:20: Resolve 260: a udp 81.26.212.150 6060 [9] 2012/09/15 00:03:20: Resolve 260: udp 81.26.212.150 6060 [5] 2012/09/15 00:03:20: SIP Tx udp:81.26.212.150:6060: SIP/2.0 200 Ok Via: SIP/2.0/UDP 81.26.212.150:6060;branch=0 From: <sip:pinger@81.26.212.150>;tag=8f105f66 To: <sip:2.228.101.190:24942>;tag=6dedcc0421 Call-ID: 4f7b936-4b5e9ec8-4d604e4@81.26.212.150 CSeq: 1 OPTIONS Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Content-Length: 0 [5] 2012/09/15 00:03:20: SIP Rx udp:81.26.212.150:6060: OPTIONS sip:2.228.101.190:24942 SIP/2.0 Via: SIP/2.0/UDP 81.26.212.150:6060;branch=0 From: sip:pinger@81.26.212.150;tag=9f105f66 To: sip:2.228.101.190:24942 Call-ID: 4f7b936-5b5e9ec8-4d604e4@81.26.212.150 CSeq: 1 OPTIONS Content-Length: 0 [9] 2012/09/15 00:03:20: Resolve 261: aaaa udp 81.26.212.150 6060 [9] 2012/09/15 00:03:20: Resolve 261: a udp 81.26.212.150 6060 [9] 2012/09/15 00:03:20: Resolve 261: udp 81.26.212.150 6060 [5] 2012/09/15 00:03:20: SIP Tx udp:81.26.212.150:6060: SIP/2.0 200 Ok Via: SIP/2.0/UDP 81.26.212.150:6060;branch=0 From: <sip:pinger@81.26.212.150>;tag=9f105f66 To: <sip:2.228.101.190:24942>;tag=2a5c9a957d Call-ID: 4f7b936-5b5e9ec8-4d604e4@81.26.212.150 CSeq: 1 OPTIONS Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Content-Length: 0 [5] 2012/09/15 00:03:20: SIP Rx udp:81.26.212.150:6060: OPTIONS sip:2.228.101.190:24942 SIP/2.0 Via: SIP/2.0/UDP 81.26.212.150:6060;branch=0 From: sip:pinger@81.26.212.150;tag=af105f66 To: sip:2.228.101.190:24942 Call-ID: 4f7b936-6b5e9ec8-4d604e4@81.26.212.150 CSeq: 1 OPTIONS Content-Length: 0 [9] 2012/09/15 00:03:20: Resolve 262: aaaa udp 81.26.212.150 6060 [9] 2012/09/15 00:03:20: Resolve 262: a udp 81.26.212.150 6060 [9] 2012/09/15 00:03:20: Resolve 262: udp 81.26.212.150 6060 [5] 2012/09/15 00:03:20: SIP Tx udp:81.26.212.150:6060: SIP/2.0 200 Ok Via: SIP/2.0/UDP 81.26.212.150:6060;branch=0 From: <sip:pinger@81.26.212.150>;tag=af105f66 To: <sip:2.228.101.190:24942>;tag=3b8e58f455 Call-ID: 4f7b936-6b5e9ec8-4d604e4@81.26.212.150 CSeq: 1 OPTIONS Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Content-Length: 0 [8] 2012/09/15 00:03:20: Play audio_en/co_welcome_conference.wav, caching false [8] 2012/09/15 00:03:20: call port 11: state code from 200 to 200 [8] 2012/09/15 00:03:22: Play audio_en/co_enter_pin.wav, caching false [8] 2012/09/15 00:03:22: call port 11: state code from 200 to 200 [5] 2012/09/15 00:03:31: SIP Rx udp:84.199.103.82:5060: REGISTER sip:sips14.s.tw SIP/2.0 Via: SIP/2.0/UDP 172.16.2.10:5060;branch=z9hG4bK4b47edca8435D139 From: "7000" <sip:7000@sips14.s.tw>;tag=2EA9980C-B2C1BC0B To: <sip:7000@sips14.s.tw> CSeq: 2667 REGISTER Call-ID: 7ecc7eb7-adc047ce-d29eb85d@172.16.2.10 Contact: <sip:7000@172.16.2.10:5060>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER" User-Agent: PolycomSoundStationIP-SSIP_7000-UA/3.2.4.0267 Accept-Language: en Authorization: Digest username="7000", realm="sips14.s.tw", nonce="642af809e14ae35f6f2b0fe0803a921f", uri="sip:sips14.s.tw", response="69ff23db49333866fb866c7c189d8995", algorithm=MD5 Max-Forwards: 70 Expires: 3600 Content-Length: 0 [9] 2012/09/15 00:03:31: Resolve 263: udp 84.199.103.82 5060 [5] 2012/09/15 00:03:31: SIP Tx udp:84.199.103.82:5060: SIP/2.0 200 Ok Via: SIP/2.0/UDP 172.16.2.10:5060;branch=z9hG4bK4b47edca8435D139;rport=5060;received=84.199.103.82 From: "7000" <sip:7000@sips14.s.tw>;tag=2EA9980C-B2C1BC0B To: <sip:7000@sips14.s.tw>;tag=370d21a9d7 Call-ID: 7ecc7eb7-adc047ce-d29eb85d@172.16.2.10 CSeq: 2667 REGISTER Contact: <sip:7000@172.16.2.10:5060>;expires=59 Server: snomONE/4.5.0.1075 Delta Aurigids Content-Length: 0 [5] 2012/09/15 00:03:40: SIP Rx udp:81.201.83.45:5060: BYE sip:8877@174.16.25.5:5060;transport=udp SIP/2.0 Call-ID: IUP7KEWC4BDSHIFRHS65MA27HI@81.201.83.45 CSeq: 103 BYE From: "39333333333" <sip:39333333333@voxbone.com>;tag=19899 To: <sip:8877@sips14.s.tw>;tag=8828815886 Via: SIP/2.0/UDP 81.201.83.45:5060;branch=z9hG4bK234e3dd4eeaa0cef7584328a76c4a662 Max-Forwards: 69 User-Agent: Vox Callcontrol Content-Length: 0 [9] 2012/09/15 00:03:40: Resolve 264: aaaa udp 81.201.83.45 5060 [9] 2012/09/15 00:03:40: Resolve 264: a udp 81.201.83.45 5060 [9] 2012/09/15 00:03:40: Resolve 264: udp 81.201.83.45 5060 [5] 2012/09/15 00:03:40: SIP Tx udp:81.201.83.45:5060: SIP/2.0 200 Ok Via: SIP/2.0/UDP 81.201.83.45:5060;branch=z9hG4bK234e3dd4eeaa0cef7584328a76c4a662 From: "39333333333" <sip:39333333333@voxbone.com>;tag=19899 To: <sip:8877@sips14.s.tw>;tag=8828815886 Call-ID: IUP7KEWC4BDSHIFRHS65MA27HI@81.201.83.45 CSeq: 103 BYE Contact: <sip:8877@174.16.25.5:5060;transport=udp> User-Agent: snomONE/4.5.0.1075 Delta Aurigids Content-Length: 0 [8] 2012/09/15 00:03:40: Clearing call port 11, SIP call id IUP7KEWC4BDSHIFRHS65MA27HI@81.201.83.45 [8] 2012/09/15 00:03:40: Remove leg 12: call port 11, SIP call id IUP7KEWC4BDSHIFRHS65MA27HI@81.201.83.45 [8] 2012/09/15 00:03:40: Hangup: Call 11 not found Quote Link to comment Share on other sites More sharing options...
global_s Posted September 15, 2012 Author Report Share Posted September 15, 2012 I had to ask my provider to change the DTMF signaling from RTP 2883 to SIP INFO to make it work again. Still don't understand why it did work before. Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted September 15, 2012 Report Share Posted September 15, 2012 Hmm. The log looks "beautiful". Sometimes the provider also change things, which can drive you nuts trying to figure out if any change on your side is the problem. The PBX also allows that in addition to listening to RFC2833 (RFC4733 now) you can also do inband DTMF detection. Maybe this is the way to go for you now. Quote Link to comment Share on other sites More sharing options...
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